Refactor webrtc_perf_tests into several source_sets.

webrtc/video/BUILD.gn

Index: webrtc/video/BUILD.gn
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index 87a9800b85f8c1a0ac484707f52629bce10829d4..54464cf8d940d289583f1a6250c864cc761b55ee 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -89,6 +89,7 @@ if (rtc_include_tests) {
       "../media:rtc_media_base",
       "../system_wrappers",
       "//testing/gtest",
+      "//webrtc/test:test_renderer",
     ]
     if (!is_android) {
       deps += [ "../modules/video_capture:video_capture_internal_impl" ]
@@ -99,6 +100,25 @@ if (rtc_include_tests) {
     }
   }
 
+  rtc_source_set("video_full_stack_tests") {
+    testonly = true
+    sources = [
+      "full_stack_tests.cc",
+    ]
+    deps = [
+      ":video_quality_test",
+      "//testing/gtest",
+      "//webrtc/test:test_common",
+    ]
+    if (!build_with_chromium && is_clang) {
+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+    if (rtc_use_h264) {
+      defines = [ "WEBRTC_USE_H264" ]
+    }
+  }
+
   rtc_executable("video_loopback") {
     testonly = true
     sources = [